FDKAAC(1)

NAME

fdkaac - command line frontend for libfdk-aac encoder

SYNOPSIS

fdkaac [OPTIONS][FILE]

DESCRIPTION

fdkaac reads linear PCM audio in either WAV, raw PCM, or CAF format, and encodes it into either M4A / AAC file.

If the input file is "-", data is read from stdin. Likewise, if the output file is "-", data is written to stdout if one of streamable AAC transport formats are selected by -f.

When CAF input and M4A output is used, tags in CAF file are copied into the resulting M4A.

OPTIONS

-h, --help
    Show command help

-o <FILE>
    Output filename.

-p, --profile <n>
    Target profile (MPEG4 audio object type, AOT)

    2   MPEG-4 AAC LC (default)

    5   MPEG-4 HE-AAC (SBR)

    29  MPEG-4 HE-AAC v2 (SBR+PS)

    23  MPEG-4 AAC LD

    39  MPEG-4 AAC ELD

    129 MPEG-2 AAC LC

    132 MPEG-2 HE-AAC (SBR)

    156 MPEG-2 HE-AAC v2 (SBR+PS)

-b, --bitrate <n>
    Target bitrate (for CBR)

-m, --bitrate-mode <n>
    Bitrate configuration mode. Available VBR quality value depends on other parameters such as profile, sample rate, or number of channels.

    0   CBR (default)

    1-5 VBR (higher value -> higher bitrate)

-w, --bandwidth <n>
    Frequency bandwidth (lowpass cut-off frequency) in Hz. Available on AAC LC only.

-a, --afterburner <n>
    Configure afterburner mode. When enabled, quality is increased at the expense of additional computational workload.

    0   Off

    1   On (default)

-L, --lowdelay-sbr <n>
    Configure SBR activity on AAC ELD.

    -1  Use ELD SBR auto configuration

    0   Disable SBR on ELD (default)

    1   Enable SBR on ELD

-s, --sbr-ratio <n>
    Controls activation of downsampled SBR.

    0   Use lib default (default)

    1   Use downsampled SBR (default for ELD+SBR)

    2   Use dual-rate SBR (default for HE-AAC)

    Dual-rate SBR is what is normally used for HE-AAC, where AAC is encoded at half the sample rate of SBR, hence "dual rate". On the other hand, downsampled SBR uses same sample rate for both of AAC and SBR (single rate), therefore downsampled SBR typically consumes more bitrate.

    Downsampled SBR is newly introduced feature in FDK encoder library version 3.4.12. When libfdk-aac in the system doesn't support this, dual-rate SBR will be used. When available, dual-rate SBR is the default for HE-AAC and downsampled SBR is the default for ELD+SBR.

    Note that downsampled HE-AAC is not so common as dual-rate one. When downsampled HE-AAC is selected, fdkaac is forced to choose explicit hierarchical SBR signaling, which (at least) iTunes doesn't accept.

-f, --transport-format <n>
    Transport format. Tagging and gapless playback is only available on M4A. Streaming to stdout is only available on others.

    0   M4A (default)

    1   ADIF

    2   ADTS

    6   LATM MCP=1

    7   LATM MCP=0

    10  LOAS/LATM (LATM within LOAS)

-C, --adts-crc-check
    Add CRC protection on ADTS header.

-h, --header-period <n>
    StreamMuxConfig/PCE repetition period in the transport layer.

-G, --gapless-mode <n>
    Method to declare amount of encoder delay (and padding) in M4A container. These values are mandatory for proper gapless playback on player side.

    0   iTunSMPB (default)

    1   ISO standard (edts and sgpd)

    2   Both

--include-sbr-delay
    When specified, count SBR decoder delay in encoder delay.

    This is not iTunes compatible and will lead to gapless playback issue on LC only decoder, but this is the default behavior of FDK library.

    Whether counting SBR decoder delay in encoder delay or not result in incompatibility in gapless playback. You should pick which one will work for your favorite player.

    However, it's better not to choose SBR at all if you want gapless playback. LC doesn't have such issues.

-I, --ignorelength
    Ignore length field of data chunk in input WAV file.

-S, --silent
    Don't print progress messages.

--moov-before-mdat
    Place moov box before mdat box in M4A container. This option might be important for some hardware players, that are known to refuse moov box placed after mdat box.

-R, --raw
    Regard input as raw PCM.

--raw-channels <n>
    Specify number of channels of raw input (default: 2)

--raw-rate <n>
    Specify sample rate of raw input (default: 44100)

--raw-format <spec>
    Specify sample format of raw input (default: "S16L"). Spec is as the following (case insensitive):

    1st char -- type of sample
        S (igned) | U (nsigned) | F (loat)

    2nd part (in digits)
        bits per channel

    Last char -- endianness (can be omitted)
        L (ittle, default) | B (ig)

--title <string>
    Set title tag.

--artist <string>
    Set artist tag.

--album <string>
    Set album tag.

--genre <string>
    Set genre tag.

--date <string>
    Set date tag.

--composer <string>
    Set composer tag.

--grouping <string>
    Set grouping tag.

--comment <string>
    Set comment tag.

--album-artist <string>
    Set album artist tag.

--track <number[/total]>
    Set track tag, with or without number of total tracks.

--disk <number[/total]>
    Set disk tag, with or without number of total discs.

--tempo <n>
    Set tempo (BPM) tag.

--tag <fcc>:<value>
    Set iTunes predefined tag with explicit fourcc key and value. See <https://code.google.com/p/mp4v2/wiki/iTunesMetadata> for known predefined keys. You can omit first char of fcc when it is the copyright sign.

--tag-from-file <fcc>:<filename>
    Same as --tag, but set content of file as tag value.

--long-tag <name>:<value>
    Set arbitrary tag as iTunes custom metadata. Stored in com.apple.iTunes field.

--tag-from-json <filename[?dot_notation]>
    Read tags from JSON. By default, tags are assumed to be direct children of the root object in JSON. Optionally you can specify arbitrary dot notation to locate the object containing tags.

EXAMPLES

Encode WAV file into a M4A file. MPEG4 AAC LC, VBR quality 3:

    fdkaac -m3 foo.wav

Encode WAV file into a M4A file. MPEG4 HE-AAC, bitrate 64kbps:

    fdkaac -p5 -b64 foo.wav

Piping from ffmpeg (you need version supporting CAF output):

    ffmpeg -i foo.flac -f caf - | fdkaac -b128 - -o foo.m4a

Import tags via json:

    ffprobe -v 0 -of json -show_format foo.flac >foo.json
    flac -dc foo.flac | \
    fdkaac - -ox.m4a -m2 --import-tag-from-json=foo.json?format.tags

NOTES

Upto 32bit integer or 64bit floating point format is supported as input. However, FDK library is implemented based on fixed point math and only supports 16bit integer PCM. Therefore, be wary of clipping. You might want to dither/noise shape beforehand when your input has higher resolution.

Following channel layouts are supported by the encoder.

1ch C

2ch L R

3ch C L R

4ch C L R Cs

5ch C L R Ls Rs

5.1ch
    C L R Ls Rs LFE

7.1ch (front)
    C Lc Rc L R Ls Rs LFE

7.1ch (rear)
    C L R Ls Rs Rls Rrs LFE

AUTHORS
nu774 <honeycomb77@gmail.com>.